moonlight-stream/moonlight-qt
1
0
Fork 0
mirror of https://github.com/moonlight-stream/moonlight-qt.git synced 2025-07-03 08:15:37 +00:00
Code Issues Packages Projects Releases Wiki Activity

Remove excess latency control logic from SDL renderer, since it doesn't appear to work very well anyway

Browse Source
This commit is contained in:
Cameron Gutman 2019-01-06 13:28:43 -08:00
parent c21ba5c808
commit 52ef84873e
2 changed files with 1 additions and 84 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
app/streaming/audio/renderers/sdl.h
View File

@ -16,9 +16,4 @@ public:
private: private:
SDL_AudioDeviceID m_AudioDevice; SDL_AudioDeviceID m_AudioDevice;
Uint32 m_ChannelCount;
Uint32 m_PendingDrops;
Uint32 m_PendingHardDrops;
Uint32 m_SampleIndex;
Uint32 m_BaselinePendingData;
}; };

80
app/streaming/audio/renderers/sdlaud.cpp
View File

@ -5,18 +5,8 @@
#include <QtGlobal> #include <QtGlobal>
#define MIN_QUEUED_FRAMES 2
#define MAX_QUEUED_FRAMES 4
#define STOP_THE_WORLD_LIMIT 20
#define DROP_RATIO_DENOM 32
SdlAudioRenderer::SdlAudioRenderer() SdlAudioRenderer::SdlAudioRenderer()
: m_AudioDevice(0), : m_AudioDevice(0)
m_ChannelCount(0),
m_PendingDrops(0),
m_PendingHardDrops(0),
m_SampleIndex(0),
m_BaselinePendingData(0)
{ {
SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO)); SDL_assert(!SDL_WasInit(SDL_INIT_AUDIO));
if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) { if (SDL_InitSubSystem(SDL_INIT_AUDIO) != 0) {
@ -61,27 +51,6 @@ bool SdlAudioRenderer::prepareForPlayback(const OPUS_MULTISTREAM_CONFIGURATION*
have.samples, have.samples,
have.size); have.size);
// SDL counts pending samples in the queued
// audio size using the WASAPI backend. This
// includes silence, which can throw off our
// pending data count. Get a baseline so we
// can exclude that data.
m_BaselinePendingData = 0;
#ifdef Q_OS_WIN32
for (int i = 0; i < 100; i++) {
m_BaselinePendingData = qMax(m_BaselinePendingData, SDL_GetQueuedAudioSize(m_AudioDevice));
SDL_Delay(10);
}
#endif
m_BaselinePendingData *= 2;
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Baseline pending audio data: %d bytes",
m_BaselinePendingData);
m_ChannelCount = opusConfig->channelCount;
m_SampleIndex = 0;
m_PendingDrops = m_PendingHardDrops = 0;
// Start playback // Start playback
SDL_PauseAudioDevice(m_AudioDevice, 0); SDL_PauseAudioDevice(m_AudioDevice, 0);
@ -102,53 +71,6 @@ SdlAudioRenderer::~SdlAudioRenderer()
bool SdlAudioRenderer::submitAudio(short* audioBuffer, int audioSize) bool SdlAudioRenderer::submitAudio(short* audioBuffer, int audioSize)
{ {
m_SampleIndex++;
Uint32 queuedAudio = SDL_GetQueuedAudioSize(m_AudioDevice);
if (queuedAudio > m_BaselinePendingData) {
queuedAudio -= m_BaselinePendingData;
}
else {
queuedAudio = 0;
}
Uint32 framesQueued = queuedAudio / (SAMPLES_PER_FRAME * m_ChannelCount * sizeof(short));
// We must check this prior to the below checks to ensure we don't
// underflow if framesQueued - m_PendingHardDrops < 0.
if (framesQueued <= MIN_QUEUED_FRAMES) {
m_PendingDrops = m_PendingHardDrops = 0;
}
// Pend enough drops to get us back to MIN_QUEUED_FRAMES, checking first
// to ensure we don't underflow.
else if (framesQueued > m_PendingHardDrops &&
framesQueued - m_PendingHardDrops > STOP_THE_WORLD_LIMIT) {
m_PendingHardDrops = framesQueued - MIN_QUEUED_FRAMES;
SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
"Pending hard drop of %u audio frames",
m_PendingHardDrops);
}
// If we're under the stop the world limit, we can drop samples
// gracefully over the next little while.
else if (framesQueued > m_PendingHardDrops + m_PendingDrops &&
framesQueued - m_PendingHardDrops - m_PendingDrops > MAX_QUEUED_FRAMES) {
m_PendingDrops = framesQueued - MIN_QUEUED_FRAMES;
}
// Determine if this frame should be dropped
if (m_PendingHardDrops != 0) {
// Hard drops happen all at once to forcefully
// resync with the source.
m_PendingHardDrops--;
return true;
}
else if (m_PendingDrops != 0 && m_SampleIndex % DROP_RATIO_DENOM == 0) {
// Normal drops are interspersed with the audio data
// to hide the glitches.
m_PendingDrops--;
return true;
}
if (SDL_QueueAudio(m_AudioDevice, audioBuffer, audioSize) < 0) { if (SDL_QueueAudio(m_AudioDevice, audioBuffer, audioSize) < 0) {
SDL_LogError(SDL_LOG_CATEGORY_APPLICATION, SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
"Failed to queue audio sample: %s", "Failed to queue audio sample: %s",